The most common call type is a call established between two parties for one-to-one communication. The standard way to set up a two-party call requires explicit control plane's signaling that allows the call parties to establish a channel where the audio data can be transferred and to negotiate the communication capabilities, for example the audio codec and the relative compression rate can be determined in this phase. Afterwards the actual voice communication can start and the audio data can be transmitted by the call parties.
Voice over Internet Protocol (VoIP) enables a speech communication over an IP connection. The Session Initiation Protocol (SIP, RFC2543), the standard protocol used for call establishment in “VoIP” based communication systems, requires some amount of signaling for each SIP session setup. In particular for two-party call an end-to-end “three round” INVITE transaction (by which a SIP session is initiated) has to be performed. The INVITE request asks the called party to join a particular two-party conversation. After the called party has agreed to participate in the call (by 200OK message), the caller confirms that it has received that response by sending an ACK request. The INVITE request typically contains a session description, for example written in Session Description Protocol (SDP, RFC2327) format that provides the called party with enough information to join the session. The session description normally enumerates the media types and formats that the caller is willing to use and where he wishes the media data to be sent. If the called party wishes to accept the call, he responds to the invitation by returning a similar description. Further, the control plane's signaling (which information cannot be lost during its transmission) and the user-plane's audio data (which may accept some loss but has real time characteristics) have different transport level requirements, which normally entail the transmission of the IP packets containing their corresponding data on separate bearers.
In some communication systems it is more important to have a fast call setup than to support end-to-end negotiation between the call parties which could even not be required at all if the audio codec and the relative parameters were fixed. Considering in particular a VoIP based environment this would mean that time consuming SIP signaling should be minimized in order to obtain fast call setup.
A mobile communications system refers generally to any telecommunications system, which enables communication when users are moving within the service area of the system. A typical mobile communications system is a Public Land Mobile Network (PLMN). Often the mobile communications network is an access network providing a user with wireless access to external networks, hosts, or services offered by specific service providers.
Professional mobile radio or private mobile radio (PMR) systems are dedicated radio systems developed primarily for professional and governmental users, such as the police, military forces, oil plants, etc. PMR services have been offered via dedicated PMR networks built with dedicated PMR technologies. This market is divided between several technologies analog, digital, conventional and trunked—none of which has a dominating role. TETRA (Terrestrial Trunked Radio) is a standard defined by ETSI (European Telecommunications Standards Institute) for digital PMR systems. U.S. Pat. No. 6,141,347 discloses a wireless communications system which uses multicast addressing and decentralized processing in group calls.
One special feature offered by the PMR systems is group communication. The term “group”, as used herein, refers to any logical group of three or more users intended to participate in the same group communication, e.g. call. Group communication with a push-to-talk feature is one of the essential features of any PMR network overcoming this problem. Generally, in group voice communication with a “push-to-talk, release-to-listen” feature, a group call is based on the use of a pressel (PTT, push-to-talk switch) in a telephone as a switch: by pressing a PTT the user indicates his desire to speak, and the user equipment sends a service request to the network. The network either rejects the request or allocates the requested resources on the basis of predetermined criteria, such as the availability of resources, priority of the requesting user, etc. At the same time, a connection is established also to all other active users in the specific subscriber group. After the voice connection has been established, the requesting user can talk and the other users listen on the channel. When the user releases the PTT, the user equipment signals a release message to the network, and the resources are released. Thus, the resources are reserved only for the actual speech transaction or speech item. One interesting advantage of the push-to-talk communication is a short call setup time, which makes the push-to-talk type of speech calls attractive to several other types of users, too.